[OpenSIPS-Users] Integration with Asterisk/Trixbox
James Lamanna
jlamanna at gmail.com
Wed May 20 22:11:59 CEST 2009
On Wed, May 20, 2009 at 7:46 AM, Iñaki Baz Castillo <ibc at aliax.net> wrote:
> 2009/5/20 James Lamanna <jlamanna at gmail.com>:
>> Hi,
>> I want to use OpenSIPs as the registrar (and NAT handler) for an
>> Asterisk/Trixbox installation.
>> I've got things partially working, but I've totally made a mess of my
>> config (I can post it if you would like).
>>
>> Some things that I need:
>>
>> I'm having problems with SIP<->SIP calls because I need asterisk to
>> stay in the media stream, so really the call has to be routed like:
>>
>> phone1 <--> opensips <--> asterisk <--> opensips <--> phone2.
>>
>> Does anyone have any configs that come close to this that I could stare at?
>
> Set "canreinvite=no" for opensips peer in sip.conf.
>
>
>> The ones I've found on the web are useful in some ways, but not in others.
>
> This question is more related to Asterisk.
Not really.
I'm using the basic NAT example with a little rewriting (as shown below).
The problem I have is that the SDP address is not being rewritten, so
on the asterisk box I see audio traces like this:
(x.x.x.x is the phone NAT IP address, y.y.y.y is the asterisk box address).
You'll notice that the first 2 lines are correct, but the second 2 are
not (asterisk sends to the private IP of the phone)
13:04:54.228511 IP (tos 0x0, ttl 118, id 8879, offset 0, flags [none],
proto: UDP (17), length: 200) x.x.x.x.47127 > y.y.y.y.12232: UDP,
length 172
13:04:54.228555 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF],
proto: UDP (17), length: 200) y.y.y.y.10536 > x.x.x.x.47128: UDP,
length 172
13:04:54.243209 IP (tos 0x0, ttl 54, id 22237, offset 0, flags
[none], proto: UDP (17), length: 200) x.x.x.x.47128 > y.y.y.y.10536:
UDP, length 172
13:04:54.243254 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF],
proto: UDP (17), length: 200) y.y.y.y.12232 > 10.20.200.219.49154:
UDP, length 172
When asterisk sends an INVITE to connect the called phone, the SDP
isn't getting rewritten for some reason.
How do I make that work?
The setup signaling looks like:
Calling phone sends INVITE --> opensips forwards to asterisk -->
asterisk sends INVITE ---> opensips forwards INVITE to called phone...
-- James
debug=3 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
# Uncomment these lines to enter debugging mode
fork=no
log_stderror=yes
#debug=4
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
children=4
listen=udp:z.z.z.z:5060
# ------------------ module loading ----------------------------------
#set module path
#mpath="/usr/local/lib/opensips/modules/"
mpath="/usr/local/lib64/opensips/modules/"
# Uncomment this if you want to use SQL database
#loadmodule "db_mysql.so"
loadmodule "sl.so"
loadmodule "tm.so"
loadmodule "signaling.so"
loadmodule "rr.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "mi_fifo.so"
loadmodule "xlog.so"
# Uncomment this if you want digest authentication
# db_mysql.so must be loaded !
#loadmodule "auth.so"
#loadmodule "auth_db.so"
# !! Nathelper
loadmodule "nathelper.so"
# ----------------- setting module-specific parameters ---------------
# -- mi_fifo params --
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
# -- usrloc params --
modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
#modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#modparam("auth_db", "password_column", "password")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# !! Nathelper
modparam("usrloc","nat_bflag",6)
modparam("nathelper","sipping_bflag",8)
modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT
modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy/rtpproxy.sock")
# ------------------------- request routing logic -------------------
# main routing logic
route{
^[
xlog("L_INFO", "New request - Request/failure/branch routes: M=$rm
RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
};
if (msg:len >= 2048 ) {
sl_send_reply("513", "Message too big");
exit;
};
if (is_method("OPTIONS")) {
sl_send_reply("200", "OK");
exit;
}
# !! Nathelper
# Special handling for NATed clients; first, NAT test is
# executed: it looks for via!=received and RFC1918 addresses
# in Contact (may fail if line-folding is used); also,
# the received test should, if completed, should check all
# vias for rpesence of received
if (nat_uac_test("3")) {
# Allow RR-ed requests, as these may indicate that
# a NAT-enabled proxy takes care of it; unless it is
# a REGISTER
if (is_method("REGISTER") || !is_present_hf("Record-Route")) {
xlog("L_INFO", "LOG:Someone trying to register from
private IP, rewriting\n");
# This will work only for user agents that support symmetric
# communication. We tested quite many of them and majority is
# smart enough to be symmetric. In some phones it takes a
# configuration option. With Cisco 7960, it is called
# NAT_Enable=Yes, with kphone it is called "symmetric media" and
# "symmetric signalling".
# Rewrite contact with source IP of signalling
fix_nated_contact();
if ( is_method("INVITE") ) {
xlog("L_INFO", "NAT: FIXING SDP");
fix_nated_sdp("1"); # Add direction=active to SDP
};
};
force_rport(); # Add rport parameter to topmost Via
setbflag(6); # Mark as NATed
# if you want sip nat pinging
# setbflag(8);
};
};
# subsequent messages withing a dialog should take the
# path determined by record-routing
if (loose_route()) {
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
route(1);
exit;
};
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
if (!is_method("REGISTER"))
record_route();
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(1);
exit;
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if (uri==myself) {
if (is_method("REGISTER")) {
# Uncomment this if you want to use digest authentication
#if (!www_authorize("siphub.org", "subscriber")) {
# www_challenge("siphub.org", "0");
# return;
#};
save("location");
exit;
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
exit;
};
if (is_method("NOTIFY")) {
route(2);
}
if(is_method("OPTIONS")) {
# send reply for each options request
sl_send_reply("200", "ok");
exit;
}
if (from_uri =~ "sip:.*@y\.y\.y\..*") {
log("request from asterisk\n");
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
exit;
};
#fix_nated_contact();
#if (is_method("INVITE"))
# fix_nated_sdp("3");
#force_rport(); # Add rport parameter to topmost Via
#setbflag(6); # Mark as NATed
} else if (is_method("INVITE")) {
#} else { #if (uri =~ "sip:1[0-9]{10,10}@z\.z\.z\.z") {
rewritehostport("y.y.y.y:5061");
#};
}
# native SIP destinations are handled using our USRLOC DB
#if (!lookup("location")) {
# sl_send_reply("404", "Not Found");
# exit;
#};
};
append_hf("P-hint: usrloc applied\r\n");
route(1);
}
route[1]
{
# !! Nathelper
if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" &&
!search("^Route:")){
sl_send_reply("479", "We don't forward to private IP addresses");
exit;
};
# if client or server know to be behind a NAT, enable relay
if (isbflagset(6)) {
force_rtp_proxy();
};
# NAT processing of replies; apply to all transactions (for example,
# re-INVITEs from public to private UA are hard to identify as
# NATed at the moment of request processing); look at replies
t_on_reply("1");
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
}
# !! Nathelper
onreply_route[1] {
# NATed transaction ?
if (isbflagset(6) && status =~ "(183)|2[0-9][0-9]") {
fix_nated_contact();
force_rtp_proxy();
# otherwise, is it a transaction behind a NAT and we did not
# know at time of request processing ? (RFC1918 contacts)
} else if (nat_uac_test("1")) {
fix_nated_contact();
};
}
route[2] {
if (!t_newtran()) {
sl_reply_error();
exit;
};
t_reply("200", "OK");
exit;
}
More information about the Users
mailing list